1515 Route Ten
Parsippany, NJ 07054
Dialogic's DM3 IP link servers are designed to enable individuals to communicate
directly over data networks in a variety of ways: phone-to-phone, PC-to-phone,
phone-to-PC, fax-to-fax, and Web browser-to-phone. Based on Dialogic's DM3 mediastream
resource architecture the DM3 IPLink family of server platforms provides support for
standard protocols, such as H.323 and G.723.1, both specified by the IMTC (International
Multimedia Teleconferencing Consortium) voice over IP forum. Furthermore, the IPLink
platform supports standard coders, such as G.729a, G.711, GSM, and Voxware RT24. It
provides an open development framework, which enables custom system configurations and
field downloads of software upgrade modules.
A recent addition to the IPLink family of products, Dialogic's DM/IP2000 is an enhanced
IP telephony platform with an integrated on-board high-speed Ethernet interface, which
results in greatly reduced latency, while increasing scalability and system reliability
for developers. Since all data/voice packets are handled directly on the board, there is a
reduction in end-to-end delays. Scalability is enhanced because the TCP/IP and H.323
stacks (which run on-board) reduce the host CPU load factor, also enabling reduced call
setup times and a smaller host software footprint. The platform has proven to be
compatible with leading H.323 applications such as Microsoft's NetMeeting, VocalTec's
Internet Phone, and Intel's Internet Video Phone.
TAP-804 DSP Resource Board
8 Centennial Drive
Peabody, MA 01960
Analogic's TAP-804 DSP resource board is an open DSP-based computer telephony processor
that is designed to deliver a full trunk's worth of audio compression - up to 30 channels
- in a single PCI slot. The board is specifically designed for advanced CTI applications,
such as Voice over IP. The TAP-804 delivers efficient system performance by taking
advantage of the fast PCI bus system, yielding greater available bandwidth for CTI
applications. Up to twelve, 60-Mhz floating point processors provide as much as 720 MFLOPS
of compute power enabling the TAP-804 to support up to 30 channels of voice compression,
and that includes G.723.1, G.729a, and SX7300.
The TAP-804 DSP resource board supports two industry standard TDM data buses: the
Multi-Vendor Interface Protocol (MVIP) and Signal Computing System Architecture (SCSA).
Through the board's TDMRouter, developers are able to port their applications to the
TAP-804 to access the entire computer telephony market. Analogic offers a comprehensive
SDK including, libraries, sample programs, and debug, load, and other utilities.
305 College Road East
Princeton, NJ 08540
Voxware is a developer of digital speech and audio technologies for IP telephony,
multimedia software, electronic devices and wireless communications. Among its core
technologies, Voxware offers MetaVoice, the company's voice-coding technology. Unlike
conventional methods which condense the audio waveform to reduce bandwidth, MetaVoice
models every element of the human voice, including resonance, pitch, timbre, timing, and
character. This allows accurate voice reproduction without distortion, while consuming low
levels of bandwidth, storage, and processing power. The MetaVoice codecs were designed
specifically for the human voice, allowing for compression ratios of up to 53:1. This
level of compression allows voice to transmit more smoothly, with less choppiness and
delay in bandwidth-constrained environments.
Another product from VoxWare is VoxPhone, a full-duplex Internet telephone that enables
users to make real-time calls over the Internet, as well as initiate full-duplex audio
conferences of up to five people without the need for a conference server. VoxPhone is
H.323 compliant. and allows users to communicate with any other H.323-compliant phone,
including Netscape Conference.
Brooktrout Technology, Inc.
410 First Avenue
Needham, MA 02194
Brooktrout's recent introduction of a complete IP telephony line of hardware and
software products includes software development kits, as well as boards, like the TR2000
series, designed to enable vendors to develop standards-based IP voice and fax solutions
in a common software and hardware environment. The TR200 series are DSP resource boards
designed to support the transport of voice and fax traffic over IP packet data networks.
Based on 32-bit DSP chips, the TR2000 provides all of the power needed to support 24
channels of speech or fax processing. A daughterboard with a digital T1 interface lowers
system costs and saves system board slots. The TR200 comes with SpeechPac voice
compression software which provides users with multiple standard vocoders, supporting
G.723.1 and G.729a in particular. These vocoders are part of a highly integrated
compression system that includes echo canceler, DTMF and VAD capability, and can be
selected at the time of the call.
The TR2000 boards support up to 24 channels of IP voice based on ITU standard vocoders
and up to 36 channels based on Lucent's efficient SX-7300 vocoder. Brooktrout boards
support both industry standard TDM buses: MVIP and SCbus, which gives users the ability to
operate with any digital network interface board that has these interfaces. To speed
development, Brooktrout provides the BTGateway Kit, which includes the TR200 IP telephony
board with 24 channels of SpeechPac software based on G.723.1, 24 channels of FaxRelay
software (for simultaneous fax over the same data network used for voice), BTStackH.323
(H.323 protocol stack), and the Netaccess PRI-ISA24 T1 network interface board.
8 Erie Drive
Natick, MA 01760
Fusion 2.0 from Natural MicroSystems, is the evolution of their Fusion platform. Recent
advances include support for the H.323 standard, enabling IP telephony gateways built with
Fusion to support calls from a wide range of IP telephony clients including Microsoft
NetMeeting, Netscape Conference, VocalTec Internet Phone, VoxWare's VoxPhone, and
NetSpeak's WebPhone. Fusion is a hardware and software platform that provides developers
of PC-based telecommunications solutions with the scalability, performance, and support
for industry standards necessary for building server-based IP telephony applications.
Fusion incorporates a broad range of vocoders, including G.723.1, G.729a, Microsoft's GSM,
and VoxWare's MetaVoice RT24 algorithm, utilized by both Microsoft and Netscape.
The Fusion IP telephony platform consists of an Alliance Generation (AG) DSP board for
PSTN network interface and media processing, an AG Realtime/2 daughtercard for real time
vocoding, and a TX Series board which handles IP routing and data protocols. Fusion
supports a full T1 in a two ISA slots, and up to four T1s in a single chassis.
Furthermore, Fusion supports Natural MicroSystems' TX3000, which enhances scalability by
providing up to four T-spans in only 5 ISA slots. Fusion also integrates with CTAccess,
Natural MicroSystems' open and extensible development environment, making it simple for
developers to quickly bring applications to market.
|Array IP-Telephony Gateway (ATG)
Array Telecom, Inc.
1120 Finch Avenue West
North York, Ontario M3J 3H7
ATG is a complete suite of software products that provides comprehensive voice and fax
communications over any IP network, including intranets and Internets. The ATG and
associated servers enable service providers and corporations to offer high-quality,
reliable telecommunications services to their clients. The Array suite provides turnkey
inbound, calling card and prepaid services, as well as normal long-distance calling and
PC-PBX capabilities. This is accomplished with client/server software on Windows NT
TELEGATE 2.0 is Array's newest generation of its Internet telephony software solution.
Enhancements to the gateway software include DSP-based voice and fax processing, using the
same codec as in previous versions. Each DSP card will support a maximum of 30
simultaneous voice conversations or fax connections. This represents a 6-fold call volume
increase over previous TELEGATE versions. Automatic switching means each channel can be
used for fax or voice on a dynamic basis. TELEGATE gateways utilizing DSPs support a
variety of interface cards, including T1, E1, ISDN PRI, analog loop, and station
interfaces. Version 2.0 provides greater flexibility with new fax capabilities including
Minute Bartering, a TELEGATE service, allows one company's gateway to be configured to
allows other organization's gateways to terminate calls into the local dialing area of the
first company's gateway for "credits." The Minute Bartering system allows
gateway owners to cooperate over the Internet, terminating calls in the local dialing
areas of each of the participants at no cost.
Tempest Data/Voice Gateway (DVG)
733 Lakefield Road
Westlake Village, CA 91361
Franklin's Tempest DVG supports conventional phone operation. Franklin's Tempest is
designed to terminate telephone voice circuits from a telco and carry voice information as
data packets over an IP-based data network. Up to 24 channels of voice traffic can be
connected to a Tempest gateway, utilizing T1 digital trunks or analog lines via FXO/FXS
ports. An Ethernet LAN port provides connection to the data network infrastructure. The
voice packets can be routed, bridged, or tunneled through another network to a remote
Tempest where the voice traffic is sent to a station, central office, or PBX.
Depending on the number of simultaneous connections expected, the Tempest can be
connected to an IP network with an access line as small as 8 Kbps or as large as a 1.5
Mbps T1. Dial into the Tempest from the PSTN or from an office PBX extension. The gateway
compresses voice to IP packets in an 8:1 ratio. The Tempest DVG supports industry standard
G.729 voice compression and encoding.
eBridge Interactive Web Response (IWR)
15236 New Greenbrier Parkway
Beaverton, OR 97006
The eBridge IWR system supplies call centers with an Internet telephony platform to
connect customers directly to live agents, where they can simultaneously talk and share
visuals over a single analog phone line. eFusion addresses the lack of QoS standards over
public networks with its DirectQuality technology, which enables a voice and data
connection over the PSTN to the call center. Call center managers would have the option of
either a low-cost connection through the Internet with variable voice quality, or a
reliable and secure connection using eFusion's DirectQuality service.
An adjunct open architecture system, the eBridge server integrates with any ACD and
supports existing IVR applications. The server also utilizes DSP technology to provide
expandability in increments of 24 (T1) or 30 (E1) ports that can expand to multiple server
systems with thousands of ports. Hardware features of the server include:
- Two or four 200 MHz Pentium Pro processors per server unit
- G.723-compliant DSP vocoder boards
- 128 MB memory, expandable to 1 GB
Software features include:
- Operation on Windows NT 4.0
- Connectivity support for H.323, T.120, SNMP, HTTP, ODBC, TCP/IP, and RTP
- Multimedia scripting through Visual Basic, Visual C++, and Java.
Rockwell Internet Gateway
Rockwell Electronic Commerce Division
300 Bauman Court
Wood Dale, IL 60191
Rockwell's Internet Gateway bridges the gap between the Internet and the call center by
enabling Web users and agents to talk and interact directly over the Internet without the
use of a telephone or a second phone line. The gateway server also gives companies an
alternate way to route inbound and outbound telephone campaigns over the Internet instead
of higher-cost networks. Rockwell's Internet Gateway server is bundled with NetSpeak's
WebPhone, a software application allowing Windows-based multimedia PCs to perform standard
telephone functions from an on-screen graphical user interface.
All parties using the system need an MCI-compatible sound card and headset. Agent
system requirements include a 120 MHz Pentium processor, 16 MB RAM, 3GB hard drive, and
LAN connection. Gateway system minimum requirements include a 166 MHz Pentium server,
Windows NT 4.0, 32 MB RAM, 3 GB hard drive, network access cards, and Dialogic's Antares
2000/50 high-speed DSP card. Caller system minimum requirements include a 486 66 MHz
processor (75 MHz to take advantage of DSP Group's TrueSpeech), Windows 3.x/95/NT, 8 MB
RAM, 5 MB free hard drive space, and a 14.4 Kbps modem with error correction.
Cisco 3600 Voice Module
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
The voice module of the Cisco 3600 series routers enables enterprises to integrate
voice, fax, and data across existing data infrastructures while reducing costs. With
Cisco's voice over IP solution, companies can off-load branch-office traffic from the PSTN
and route is over the company's intranet, eliminating toll charges. The voice modules
interface with legacy phones, fax machines, key systems, and PBXs, making the network
telephony process transparent to users.
Voice processing latency is minimized by Cisco's router-based design and quality of
services features in Cisco IOS software, such as Resource Reservation Protocol (RSVP) and
Weighted Fair Queuing (WFQ). The dedicated DSP architecture provides processing power for
voice compression, echo cancellation, silence suppression, and jitter buffer management
without burdening the router. The voice modules support toll-quality compression scheme
such as G.711 for high-bit rate applications and G.729 for WAN applications. WAN modules
support ISDN (BRI and PRI), integrated digital modems, and T1/E1. LAN modules support
Ethernet, Fast Ethernet, and Token Ring. The framework within the IOS software also offers
support for H.323 for interoperability with third-party telephony software. The vocoder
can compress standard 64 K voice down to 8 K. List prices for the voice modules start at
$625 per port.
WebPhone Gateway Exchange (WGX) Server
902 Clint Moore Road
Boca Raton, FL 33487
The NetSpeak's WGX server brings calls placed from conventional telephones into an
IP-based data network, and calls placed using a WebPhone application into an existing
telephone system. The WebPhone application provides toll-quality, point-to-point
communication over TCP/IP networks. In order for WebPhone users to communicate with
someone using a traditional phone on the PSTN, the WGX server is needed.
The WGX server consists of hardware and software that run on the Windows NT OS. Two
industry standard interfaces are supported - conventional circuit-switched telephone
systems and data network systems. The server takes audio and signaling information from
the PSTN, compresses or decompresses it, and delivers it to the TCP/IP network. Two
interfaces are supported to the PSTN: analog, standard two-wire loop connecting directly
to a line from a telco or a line to an existing PBX; and a T1, high speed, digital
interface for connecting directly to a telco central office, PBX, or ACD. Analog and
digital boards from Natural MicroSystems are used by NetSpeak. NetSpeak also uses the RSA
Public Key Cryptosystem to provide secure communications.
VocalTec Telephony Gateway 3.1
VocalTec Communications, Inc.
35 Industrial Parkway
Northvale, NJ 07647
Building on VocalTec's Internet Phone technology, the Telephony Gateway bridges the gap
between the PSTN and the Internet or intranets to enable long-distance calling from
phone-to-phone, fax-to-fax, PC-to-phone, phone-to-PC, and Web browser-to-phone. The
VocalTec Telephony Gateway uses full-duplex echo-canceling algorithms and a management
system which optimizes audio behavior over different types of networks, and the gateway
reconstructs packets which may have been delayed or lost during transmission.
Other features include:
- Simple Dialing Plan: The Gateway auto attendant asks the caller for the destination
phone number, and the caller enters the number just as they ordinarily would. The local
Gateway automatically determines which remote Gateway should handle the call.
- Call Progress Detection: An auto detection mechanism prevents disconnects during long
periods of silence, ensuring conversations are not prematurely cut off.
- Interactive Voice Response: IVR serves as the interface between the PSTN/PBX and the
Internet or intranet. Voice prompts make the calling process user-friendly, and the system
can be customized so prompts can be recorded in any language.
- T1/E1 capability.
The VocalTec Telephony Gateway is designed for easy management. An intuitive graphical
user interface GUI), security functions, and third-party billing support are standard. The
GUI allows viewing of group connectivity, and each line can be configured separately as an
internal extension, analog line, or digital trunk line with information for up to 30 voice
and 8 fax lines.
Internet Telephony Server
Lucent Technologies, Inc.
211 Mt. Airy Road
Basking Ridge, NJ 07920
Lucent's Internet Telephony Server (ITS) is a server-based solution that places voice
and fax calls over TCP/IP networks using voice compression software developed by Bell
Labs' new division, elemedia. ITS supports industry standard codecs, including those
provided as part of the H.323 protocol stack. ITS works with DEFINITY ECS and most
existing phone systems and is connected to the PBX via a T1/E1/PRI tie line interface. The
ITS is based on an industry-standard components, including:
- Intel-based Pentium server
- Windows NT 4.0 OS
- Administration from NT screens or web browser
- Standard NICs: Ethernet 10Mb/s and 100Mb/s
- Standard DSP cards
- Standard T1/E1/PRI or analog loop-start interface
- H.323 protocol
ITS connects a company's switches or PBXs together and routes calls over the Internet
or an intranet, or the customer private or PSTN networks, depending on customer
pre-configured criteria. Up to 24 sessions can take place simultaneously, each including
either a single voice or fax call, per each T1 card in the ITS. The ITS uses a dynamic and
transparent routing algorithm for its operation (no routing decisions on the part of the
customers are needed). In addition, if the quality of the IP network falls below
pre-specified threshold, backup to PSTN is automatic.
Clarent Gateway 2.0
Redwood City, CA 94063
The Clarent Gateway 2.0 is a suite of software programs that converts a telephone
conversations and fax into digital information and compresses it 3:1 for transfer over the
Internet or TCP/IP intranets. This version brings users improved telephony connections,
enhanced scalability, and expanded management features that making billing and reporting
easier. Applications developed in conjunction with Voxware, a leader in voice compression,
squeeze more calls into less bandwidth - from 4-24 calls per gateway.
Clarent's gateway is powered by single or dual 200 MHz Pentium Pro processors, runs on
Windows NT 4.0, and requires 64 MB RAM. The telephone interface cards are Natural
MicroSystems', and Clarent supports analog, T1, E1, and ISDN connections. New features
offered by version 2.0 include real-time fax, more sophisticated management capabilities
(such as RSA encryption of customer information), redundant architecture for fault
tolerance, Simple Network Management Protocol (SNMP) support, and support for SS7 and HP
TTM & Associates, Inc.
271 East Imperial Highway
Fullerton, CA 92635
TTM's iGate is a combination Internet gateway, voice service bureau, and unified
messaging solution that uses the Internet to deliver telephone conversations in real-time
will toll-quality voice. The iGate server is accessed from any touch-tone phone. TTM
addresses Internet toll-bypass as a business application. Online prepaid calling cards and
credit card payments are integrated into the server.
Two services are incorporated into the iGate server: iCall and iOffice. iCall features
are services normally associated with a voice mail system, such as "follow me"
service, voice mail, and pager notification. iOffice uses an automated attendant, multiple
extensions, individual voice announcements, and "fax through" integration. Both
services can use debit and credit card purchasing through a Web interface and via
iGate servers are available in configuration from 8 to 100 ports per server. Multiple
servers can be combined to create any size system. TTM is a development partner with
Natural MicroSystems, and uses Fusion in creating its servers. TTM will modify the iGate
servers for custom gateway applications on a project by project basis.
LinkNet IP Telephony Gateway
140 Sherman Street
Fairfield, CT 06430
The LinkNet IP Telephony Gateway is designed as an open-system platform with a full set
of enabling technologies supporting a variety of IP communication applications, including:
- Phone to phone,
- PC to phone,
- Phone to PC,
- Real-time and Store-and-Forward IP Fax,
- Voice over Frame Relay, ATM, and Cable,
- SS7 integration.
At the heart of the hardware architecture, developers will find the
foundation-switching card, a carrier card that interfaces with the host bus, providing
connectivity to internal computer telephony bus structures (MVIP, SCSA). Current cards are
available for ISA bus and Sbus, with CompactPCI and VME implementations planned for the
coming year. Most recently, Linkon has announced a gateway for Sun Microsystems' PCI-based
Netra T 1100 and Sun Enterprise 450 computing platforms. The new card is designed to run
under the Solaris 2.6 operating system, provides data transfer rate of up to 8 Mbps, and
support for up to 48 channels per voice slot.
Based on Linkon's universal port Maestro System technology, the gateway also offers
H.323 support, which comprises data format, call control, and both audio and video
compression. At present, the LinkNet IP Telephony Gateway supports from 4 analog ports up
to multiple E1/T1 digital circuits, thus providing up to 90/96 channels for full-duplex
digital communications. According to the company, the product is designed to address the
most difficult technical issues of IP telephony, namely full-duplex communications,
scalability, echo cancellation, efficient audio compression, and low latency.
Hicom Xpress Telephony Internet Server
Siemens Business Communication Systems, Inc.
4900 Old Ironsides Drive
Santa Clara, CA 95054
Siemens Business Communication Systems' Hicom Xpress Telephony Internet Server (TIS),
enables end users to place voice and real-time fax calls over a corporate intranet or the
Internet. Siemens' TIS, incorporating components from Vienna Systems Corporation, allows
voice and real-time fax calls initiated from a PBX to be routed over an IP network.
Siemens also plans to provide phone-to-phone, real-time fax-to-fax, PC-to-phone,
phone-to-PC, and PC-to-PC communications over the Intranet/Internet infrastructure.
Transparent access to the IP trunks of Siemens' new Hicom Xpress TIS is made possible
by the automatic route selection software on Siemens' Hicom 300 E communications server.
The Siemens Hicom Xpress TIS is fully non-blocking and can handle up to 46 calls
simultaneously. It delivers toll-quality voice with an end-to-end delay of only 100 msec
and a greatly reduced bandwidth CODEC (coder/decoder) using only 7.3 Kbps per voice
channel. The product supports standard T1/PRI interfaces to a PBX and most of the popular
LAN interfaces to the IP network. E1 is anticipated for delivery in early 1998. Delivering
real-time fax service on any port on a call-by-call basis, the Universal Voice/Fax Line
feature of the TIS allows any line to start out as a voice call and switch to fax. PSTN
fallback, the ability to reroute calls back through the PBX transparently, is supported
when the IP network is congested at the time of call setup.
Access Plus F200ip
10445 Pacific Center Court
San Diego, CA 92121
Supporting phone to phone connections over frame relay, Internet, and private IP
networks, Nuera's Access Plus F200ip integrated Voice over IP gateway and Voice/Frame
Relay Access Device (FRAD) features E-CELP (Enhanced Code Excited Linear Prediction) along
with Ethernet-based IP transmission and serial frame relay trunks supporting T1/E1
throughput. The F200ip offers software configurable analog voice interfaces, digital T1/E1
PBX interfaces, and capacity of up to 30 voice and fax channels per unit for users.
Nuera's F200ip routes calls based on either user selection or system manager
As for scalability, Ethernet support allows F200ip units to be scaled up to
configurations supporting from tens to hundreds of T1/E1 connections and thousands of
voice channels as packet voice applications grow. The product includes an embedded SNMP
(Simple Network Management Protocol) agent that can be managed under any SNMP-based
manager as well as Nuera's HP OpenView for Windows workstation manager. An F-Series
configurator is designed to facilitate installation and configuration of the various Nuera
F-Series products in public and private network applications.
V/IP Phone/Fax IP Gateway
MICOM Communications, Inc.
4100 Los Angeles Avenue
Simi Valley, CA 93063
MICOM'S V/IP (Voice over IP) product family allows companies to digitize, compress, and
route intra-company voice and fax traffic over their enterprise IP networks. V/IP connects
to PBX and key telephone systems via a comprehensive line of analog (E&M, FXS, FXO)
and digital (T1, E1) voice interface cards (VICs), expandable from 4 to 24 channels for T1
and to 30 channels for E1. V/IP installs in any shared 486, Pentium, or similar PC with
NetWare, MS-DOS, Windows 95, or Windows NT. MICOM's offering uses very little processing
power, allowing the PC to be used for other tasks.
V/IP uses MICOM's ClearVoice technology with ITU standard G.729 voice compression,
silence suppression, background noise regeneration, echo cancellation, and robust voice
switching. ClearVoice runs on the VIC's 40-MIPS digital signal processors (DSPs),
providing toll-quality voice with low bandwidth requirements. In fact, V/IP connections
average less than 3 percent of a 56/64K WAN link. V/IP converts a voice call into a
digital signal at 8 Kbps. IP overhead increases this to 17 Kbps. MICOM's silence
suppression technology (which only uses bandwidth when a person is talking) then reduces
the voice signal to an average of 7 Kbps. As for flexibility, V/IP Gateways transmit over
any LAN/WAN via Ethernet, FDDI, frame relay, or ATM.
4800 Great America Parkway, Suite 400
Santa Clara, CA 95054
InnoMedia's InfoTalk Internet telephony device reroutes standard long-distance calls
over the Internet without requiring a PC. Designed for an individual user and based on
InnoMedia's patent-pending compression technology the product plugs in between the handset
and the wall connection. After establishing a normal phone connection, all the InfoTalk
user needs to decide if they want to pay for the long-distance call or use the Internet
InnoMedia has also recently announced InfoGate EP and InfoGate IP - a pair of Internet
gateways for voice and fax applications, designed for small- to medium-sized corporations.
Each gateway is an internal PC card that allows up to four simultaneous users to call
long-distance over the Internet to any destination using a standard phone. The gateways
offer the following features:
- Four PSTN lines (EP version) for local call-in and dial-out capabilities;
- Four internal extensions (IP version) allowing business users to realize cost saving
- Four-card capacity per system, allowing up to 16 ports in a single PC system;
- Backup ISDN connection ensures high-quality, low-cost voice connection during peak
Internet traffic periods.
InfoGate offers dynamic least-cost routing, full-duplex voice echo cancellation,
interactive voice response, and is compliant with the ITU's H.323 standard.
Vienna Systems Corporation
555 Legget Drive, Suite 400
Canada K2K 2X3
The Vienna.way Gateway allows users to place voice and fax calls over the Internet,
private IP networks, or the PSTN using conventional phones and fax machines. Key features
of the product include echo cancellation, compression, and transport optimized for IP
networks; PBX functionality at the desktop, such as hold, transfer, forward, and
conference; real-time fax transmission and immediate confirmation; scalable system that
can handle up to 2,000 simultaneous calls and 800 trunks; and centralized network
The Vienna.way Gateway consists of Network Interface Cards (NICs) and Digital Signal
Processor (DSP) cards installed in either a standard PCI/ISA tower or a resilient
rackmount chassis for high availability applications. The Gateway interfaces directly to
telephone networks or PBXs through either ISDN BRI or T1 interfaces. Programmable DSP
cards ensure that voice processing and packetization occur directly on the card, leaving
Gateway resources available for call processing functions such as billing, routing, and
Vienna's my.way client application turns a multimedia PC with microphone and sound card
into a multi-line phone with PBX functionality. Users can simultaneously access their
corporate voice and data networks, attend virtual meetings, and collaborate on documents
with their coworkers through a 28.8 Kbps or faster dial-up connection from anywhere in the
world. Coworkers at the main office can reach remote workers by simply dialing an
Vocal'Net IP Telephony Gateway
120 North 44th Street, Suite 200
Phoenix, AZ 85034
Using Internet Protocol (IP) Inter-Tel's Vocal'Net IP Telephony Gateway serves as a
bridge between the PSTN or PABX and an IP data network. It enables organizations to
communicate full duplex over IP networks such as the Internet or a private intranet for
potential cost savings compared to standard long-distance phone service. The Gateway
converts voice to data which consumes less bandwidth. The Vocal'Net Telephony Gateway is
scalable from 8 to hundreds of ports depending on the needs of small to large businesses.
Packetization and real-time compression take place on high-performance DSP cards, with no
significant impact on the CPU. Minimum platform requirements for the Vocal'Net include
Microsoft Windows NT operating system, a Pentium 100 processor, 10 MB of disk space, and
16 MB of RAM. The system works with any manufacturer's PBX as well as any existing
The current version of Vocal'Net allows local or remote configuration over the IP data
network for programming multiple locations worldwide, and also offers a call accounting
package (InsideTrack), which provides the ability to define and run custom reports as well
as several predefined reports. Future enhancements to Inter-Tel's telephony gateway
- Increased capacity of up to 96 ports per server;
- Support for H.323, G.723.1, G.729a, Voxware, and elemedia;
- Fax gateway, fax broadcast, and video gateway.
Network Exchange Vodex Voice Gateway
13595 Dulles Technology Drive
Herndon, VA 20171
NETRIX has announced Vodex Voice Gateway software for its Network Exchange 2210, a
Voice/Data/Fax over IP Gateway Switch. Vodex simultaneously delivers high-quality voice
over IP and voice over frame relay with the ability to gateway between the two. The
Network Exchange 2210 is a multi-service switching platform that combines WAN switching,
switched compressed voice, and multi-protocol support in a single platform. The Network
Exchange connects between the corporate PBX using either a DS1 or E1 interface and the LAN
using a standard Ethernet interface. The existing LAN carries voice/data/fax traffic
between the PBX and the egress to the WAN. Along the way, voice traffic is compressed down
to 5.5 Kbps from 64 Kbps. Each Network Exchange has one IP address per box, greatly
reducing network administration. Multiple units can be deployed within the enterprise to
handle PBXs, analog handsets, and even ISDN-based PBXs.
The functionality of the Vodex gateway is transparent to network users, helping to
ensure its use and realization of the associated cost-savings. The Network Exchange can
adapt to any dialing plan and reduce administrative costs typically associated with an
off-premise extension, a tie line or special inter-location numbering plans. The Network
Exchange can operate with any WAN technology, such as public Frame Relay or ATM services,
leased lines, satellites, and IP Virtual Private Network (VPN) services. The extensive
statistics-gathering capabilities of the Gateway provide real-time accounting information
that is useful for management and billing.
Internet Phone Fax Gateway
MicroCall (US office)
160 East 56th Street
New York, NY 10022
MicroCall's Internet Phone Fax Gateway (IPFG) is designed to deliver real-time voice
and fax communications over IP networks. Developed by IBM-Haifa Research Lab, the IPFG is
based on MicroCall's hardware and software. The Internet Phone Fax Gateway utilizes
industry standard protocols: TCP, RTP on UDP, H.323, and G.711 to provide excellent voice
quality, minimum delays, and instant connectivity at greatly reduced cost. The IPFG is
available in 8- and 16-port analog configurations, as well as T1 and E1 configurations
(24/30 ports, respectively). Other PSTN interfaces include 2-wire loop start, 2-wire
E&M, and ISDN PRI/BRI.
The gateway's billing management functions include a full range of features such as:
number of calling subscriber; number of called subscriber; length of call; and time and
date. An RS232 interface is also available to interface with external billing systems.
Special features of the gateway include:
- Built-in IVR system;
- Individual voice mailbox;
- PABX-like features, including call forward, hold, and conference;
- Unified messaging.
Network management features include local and remote management of the gateway, as well
as a network interface via SNMP.